similar to: Newbie on VoIP

Hi All, I have an asterisk box on my DMZ, and I'm using a PF for my firewall, I can make a call but some reasons I have a dead air. Any Ideas? below are my rules... ext_if = "bce0" int_if = "bce1" altitude = "172.16.1.0/24" #### machines #### vbox = "172.16.1.1" uci = "172.16.1.4" voices = "203.172.x.1" ipc =

I upgraded dovecot from v1 to v2 as part of a CentOS upgrade (5.6->6.5) and am now running on 2.0.9. Any time I connect to the pop or imap ports however I get "master: Error: service(imap-login): child 2860 killed with signal 11 (core not dumped - add -D parameter to service imap-login { executable }" after following the core dump instructions at http://www.dovecot.org/bugreport.html.

Hi All, The only Internet connection our LAN have is via a proxy server (remotely located) which is in ZONE is defined as prox and in HOSTS defined as prox eth0:<proxy_ip> Local Area Network is defined in ZONE as net and HOSTS defined as eth0:<ip_subnet> Is it possible for shorewall to be a proxy server of the local area network (net)? and shorewall will then get the connection

El Wed, 8 Jul 2020 12:23:42 -0700 John Kiniston <johnkiniston at gmail.com> escribió: > Dovid, You could use func_odb + a ODBC Redis driver to keep from having to > shell out. > > DO you know odbc redis drivers? It would be nice to store cdrs ans other stuff in redis without patching asterisk -- PekePBX, the multitenant PBX solution https://pekepbx.com

I currently have: Telco-PRI ---- Panasonic DBS576 PBX ---- E&M wink T1 ---- Asterisk. I have configured the Panasonic to forward my Asterisk DIDs to the Asterisk extensions over the T1. I do not get DID nor CID on the Asterisk, so I want to use PRI between the PBXs. I do not want to pay for another PRI card for the Panasonic. (T1 and PRI are different cards) I see this as my least

I am trying to tweak my dial plan and I am running into a problem. Sometimes my VoIP out bound calls do not complete on overseas calls(busy or just a hang-up). Is there a way in the dial plan to automatically dial out of my PRI when something like this happens. Either by time limit by a failure event? Any point in the right direction would be great Thanks, CLI output (cleansed to protect the

Someone has hacked into our system and is making calls overseas. How can I: 1. Find out the where the calls are originating from? 2. Block all calls that are not authorized? Our system is in the USA. Only calls from inside our LAN are allowed. Thank you, Gary Kuznitz -------------- next part -------------- An HTML attachment was scrubbed... URL:

Same here. I called them yesterday plus email and still no reply. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of brett-asterisk@worldcall.net Sent: Friday, January 21, 2005 10:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Webmin Module for Asterisk (and

Anyone used any of the Uniden VoIP gear, any comments one how easy it was to integrate with asterisk? Also quality of the gear in general? This unit _looks_ pretty nice on paper http://www.uniden.com/productpop/00_productpop.cfm?prd_code=UIP1868 -- Harondel J. Sibble Sibble Computer Consulting Creating solutions for the small business and home computer user. help@pdscc.com (use pgp keyid

Is anyone out there using Druid? After the switchbox announcement today I've been looking into some other gui's and as I'll probably do a trial install this weekend of the free switchvox iso but I thought I'd ask is there any other guis I should be burning trial ISO's of as well? Regards, Dean Collins Cognation Pty Ltd dean at cognation.net <mailto:dean at

On 30 April 2017 at 16:54, Tech Support <asterisk at voipbusiness.us> wrote: > I thought this was a non-commercial list. > > Yeah, I wouldn't mind so much if it had actually answered the original poster's query. "Switch to our proprietary solution and we can offer you this proprietary solution" isn't a contribution, it's an ad. -Barry > >

Hi I have a small question on CDR Database: It's used by billing software no ? he have only one structure of data or they have multi structure with more information logged ? sample: cdr simple and cdr_extended thanks bye

Hello. Well I now have asterisk installed. I've printed out the asterisk web site. I've printed the draft Asterisk handbook V2 I've printed the Introduction to the asterisk open source pbx Because I'm experimenting, I would like to do things in a certain order :- 1) VOIP inside the private LAN from one computer to another. e.g. 192.168.1.1 to 192.168.1.2 etc. 2) VOIP

Hi , I know it's might not the right way to asking such stupid question. But I want to take help from experts into VoIP fields so I have to decided to post here. Please help me which will be the best VoIP conferencing phone which will cover 10 Persians into conferencing with best audio support ? -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -------------- next

@Olivior I agree that seting up WebRTC is hard, however when done it is smooth to use. For replication you can build RPMs with working configurations. Regarding stability, it is not being used widly, so can't say it is mature. However we have no complain so far regarding audio or connectivity. sometime we provide support for "allow media / mic" type issues, but you know it is

Hello, almost any useful CLI command has an analogue on Asterisk Manager Interface, but I cannot find a way to get the list of subscriptions using AMI. Which is the command, if any? The CLI command is "sip show subscriptions" Leandro -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I haven't seen any mention of this on the list. I'm curious if anyone has tried it and can share some opinions on it? http://www.thirdlane.com/screenshots.htm http://www.thirdlane.com/opensource.htm#manager Defaults Manager - initial PBX configuration Device Manager - management of devices (phones) Mailbox Manager - configuration of user mailboxes Extensions Manager - dialplan

Sherwood, Your intentions are noble and your desire to build this, fullfills an immediate need for business. If your intention is just to build a GUI for Asterisk, read no further. If your desire is to build something more purposeful, your best bet would be to see the existing commercial GUI/HostedPBX offerings like Pbxware and Switchware from bicomsystems.com ( http://www.bicomsystems.com)

The solution you choose should be based on many factors which should include your business requirements, team's experience, your budget, growth expectations and more. You can choose Asterisk or Freeswitch as a platform and start building on that - but it is not simple and being new to VoIP you are likely to make mistakes. The "do-it-yourself" approach will some money initially, but

Thirdlane Connect can be used as a softphone. It works in modern browsers (no installation is required), on Mac, Windows and Linux desktops, and on mobile phones. Besides basic softphone functionality, it provides instant messaging, group chat (channels), voice and video conferencing, and screen sharing. It integrates with a variety of applications and CRMs such as Salesforce, Zoho, Zendesk,